From 21dac6c05ddff1ce130843a82127618f9ed766a7 Mon Sep 17 00:00:00 2001
From: Denis Krjuchkov <denis@crazydev.net>
Date: Sun, 5 May 2013 18:19:04 +0600
Subject: decoder/FLAC*: rename files and symbols to Flac*

---
 src/decoder/FlacPcm.cxx | 110 ++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 110 insertions(+)
 create mode 100644 src/decoder/FlacPcm.cxx

(limited to 'src/decoder/FlacPcm.cxx')

diff --git a/src/decoder/FlacPcm.cxx b/src/decoder/FlacPcm.cxx
new file mode 100644
index 000000000..82896bd24
--- /dev/null
+++ b/src/decoder/FlacPcm.cxx
@@ -0,0 +1,110 @@
+/*
+ * Copyright (C) 2003-2012 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "FlacPcm.hxx"
+
+#include <assert.h>
+
+static void flac_convert_stereo16(int16_t *dest,
+				  const FLAC__int32 * const buf[],
+				  unsigned int position, unsigned int end)
+{
+	for (; position < end; ++position) {
+		*dest++ = buf[0][position];
+		*dest++ = buf[1][position];
+	}
+}
+
+static void
+flac_convert_16(int16_t *dest,
+		unsigned int num_channels,
+		const FLAC__int32 * const buf[],
+		unsigned int position, unsigned int end)
+{
+	unsigned int c_chan;
+
+	for (; position < end; ++position)
+		for (c_chan = 0; c_chan < num_channels; c_chan++)
+			*dest++ = buf[c_chan][position];
+}
+
+/**
+ * Note: this function also handles 24 bit files!
+ */
+static void
+flac_convert_32(int32_t *dest,
+		unsigned int num_channels,
+		const FLAC__int32 * const buf[],
+		unsigned int position, unsigned int end)
+{
+	unsigned int c_chan;
+
+	for (; position < end; ++position)
+		for (c_chan = 0; c_chan < num_channels; c_chan++)
+			*dest++ = buf[c_chan][position];
+}
+
+static void
+flac_convert_8(int8_t *dest,
+	       unsigned int num_channels,
+	       const FLAC__int32 * const buf[],
+	       unsigned int position, unsigned int end)
+{
+	unsigned int c_chan;
+
+	for (; position < end; ++position)
+		for (c_chan = 0; c_chan < num_channels; c_chan++)
+			*dest++ = buf[c_chan][position];
+}
+
+void
+flac_convert(void *dest,
+	     unsigned int num_channels, enum sample_format sample_format,
+	     const FLAC__int32 *const buf[],
+	     unsigned int position, unsigned int end)
+{
+	switch (sample_format) {
+	case SAMPLE_FORMAT_S16:
+		if (num_channels == 2)
+			flac_convert_stereo16((int16_t*)dest, buf,
+					      position, end);
+		else
+			flac_convert_16((int16_t*)dest, num_channels, buf,
+					position, end);
+		break;
+
+	case SAMPLE_FORMAT_S24_P32:
+	case SAMPLE_FORMAT_S32:
+		flac_convert_32((int32_t*)dest, num_channels, buf,
+				position, end);
+		break;
+
+	case SAMPLE_FORMAT_S8:
+		flac_convert_8((int8_t*)dest, num_channels, buf,
+			       position, end);
+		break;
+
+	case SAMPLE_FORMAT_FLOAT:
+	case SAMPLE_FORMAT_DSD:
+	case SAMPLE_FORMAT_UNDEFINED:
+		/* unreachable */
+		assert(false);
+	}
+}
-- 
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